I run an Asterisk server and I was wondering if anyone has an up to date list of current ATA hardware that supports pulse dialing.
I have a Grandstream HT-486 (doesn't work on pulse), an HT-503 (also doesn't work), and a Digium IAXy (works!).
I would like to be able to hook up my phones 1 to 1, or at worst 2 to 1 on ATA's. I have read that pulse works on the Grandstream HT-502, but I have also read it works with the 503, which it doesn't. Unfortunately the Digium IAXy (S101) isn't made anymore.
When I search for pulse compatibility it seems all the results are 3 - 5 years old and we know what 3 - 5 years means in the VoIP world :)
The Grandstream HT-502 accepts pulse dialling but the newer devices don't. I don't know of any other new devices that support pulse dialling and I don'really expect any to be introduced.
Jack
Are you using your asterisk between your dial phones and a POTS line? If so, care to share your extensions.conf?
If your P.C. has a PCI slot, you could buy a 1 or 2 port Ds1 card (t1) card and a Adit 600 with FSX cards. This would provide 24 Stations TT/Dp. On Ebay you can find the t1 card for about $85-125 and the adit 600 for $10-$20. I've bought fully loaded Adit 600 for $10 + $18 shipping.
This will provide 100 x 's better results that the ATA's you find.
This may be more involved than you want but a idea.
Seach for T1 under Asterisk.
Cliff,
No more POTS lines here at all, all SIP trunks. Are you using "pure" Asterisk or something like FreePBX?
No PCI slot, or any slots :) I migrated my * server to a Raspberry pi about a year ago. I could accomplish what you're talking about with a SIP to T1 gateway like I installed a couple months ago for a customer, but those things are still a bit pricey! As luck would have it, I have to order a 2 port ATA for a customer, so I'll at least be able to test a 502 and make sure it works :) You'd think a 503 would work if a 502 does, but mine doesn't :(
Quote from: mdodds on April 14, 2015, 04:24:36 PM
Cliff,
No more POTS lines here at all, all SIP trunks. Are you using "pure" Asterisk or something like FreePBX?
I'm running "pure" Asterisk, v. 1.8.32.2 on FreeBSD 10.1.
I had a working setup a few years ago just using a Motorola chipset based modem as a clone X100P FXO and a soft phone on my Windows PC. I was mainly using it as an answering machine, but I think I had it configured so that I could answer incoming calls with the soft phone, and possibly so that I could make outgoing calls too. For some reason, I shelved it. I thought I still had the setup but I must have reused the hard drive for another project.
This time around, I bought a used Digium TDM400P with an FXO module included. I plan to get up to 3 FXS modules to plug vintage phones into. I am under the impression that I should be able to support pulse dialing with this hardware. I'll probably start out with 1 and see how it goes.
I have a number of ideas on things I'd like to try with it, like capturing the 10 digit number, deciding if it's long distance or not, and if long distance connect to my calling card access number and place the call through it, otherwise just make a local call. Of course I will get the voicemail feature going again. I have recorded an outgoing message with a SIT tone at the beginning hoping to fool autodialers into thinking my line is out of service. I tried this idea briefly in my previous asterisk implementation but I don't think I used it long enough to determine if it really worked. I'm also thinking of trying to interface a dial-less D1 or candlestick to the audio ports on my Windows system to use as input/output to the soft phone.
I might not actually get much of this done until next winter though, as the transition from winter to spring is progressing here and I will probably soon be spending more of my spare time outdoors. :)
I bet you'd find that using one of the Asterisk distros that use FreePBX will be a LOT easier, particularly with some of the fancier stuff you want to do. Asterisk is up to version13 now (which is actually 1.13) and if you were using zaptel for your analog cards it has now gone away too, replaced by DAHDI.
FreePBX makes it a cakewalk to do things like you were talking about with call routing depending on the number you dialled etc.
There is PBXinaFlash, Elastix, AsteriskNOW and even FreePBX themselves have a native distro.
As far as I know the Digium TDM cards handle pulse dialling just fine.
Quote from: mdodds on April 14, 2015, 10:46:17 PM
I bet you'd find that using one of the Asterisk distros that use FreePBX will be a LOT easier, particularly with some of the fancier stuff you want to do. Asterisk is up to version13 now (which is actually 1.13) and if you were using zaptel for your analog cards it has now gone away too, replaced by DAHDI.
FreePBX makes it a cakewalk to do things like you were talking about with call routing depending on the number you dialled etc.
There is PBXinaFlash, Elastix, AsteriskNOW and even FreePBX themselves have a native distro.
As far as I know the Digium TDM cards handle pulse dialling just fine.
I am running mine on older hardware so prefer a minimalistic OS installation with only the packages I need to get the job done. I haven't even put X on it and probably won't. I am using asterisk 1.8.32 because that version is available as a binary package for FreeBSD 10.1. It also has a binary package for dahdi 2.4 so I'm using that. Asterisk 1.8.32.2 was released on Jan. 28 2015 so it's only about 2-1/2 months old. It looks like they have changed their version numbering scheme so that every little bug fix jumps a major number. If these versions work for me I see no reason to update them from source, but I will if I need to. As for asking to see other's configs, I probably don't really need help, just thought it would save me a bit of the time I would otherwise spend figuring it out.
Actually they have been fairly consistent with their versioning except for that jump from 1.8 to 10....not sure whatever happened to 1.9 :) Just add 1. in front of their current releases and you get the REAL version. One thing to note is that 1.8x goes EOL this October.
What type of hardware are you running it on?
Quote from: mdodds on April 15, 2015, 02:56:04 PMWhat type of hardware are you running it on?
It's a Pentium-M 1.86 ghz on a socket 479 mini-ITX motherboard. I chose this setup for the lower power consumption and thermal characteristics such that I could run without a cpu fan. I got the biggest solid copper heatsink I could find that fit in the case. I'm not crazy about the case, but I needed one that had one full height PCI slot and didn't need a riser card for a sideways slot. I'm thinking maybe next winter I will 'update' it with a Core 2 duo mobile. :)
Quote from: mdodds on April 15, 2015, 02:56:04 PM
Actually they have been fairly consistent with their versioning except for that jump from 1.8 to 10....not sure whatever happened to 1.9 :)
That is also consistent with prior habits. Until the time of the jump to version 10, the odd numbered versions were development versions in the source code control system.
Released were only the even versions, and upon a release the version was incremented for the new development branch.
Therefore 1.9 was the development branch for 10.
The reason for the change in numbering was that there were some discussions whether Asterisk could possibly every morph into a version 2 system and what it possibly could mean in terms of features. If I recall, it was Marc S. who felt that this would never happen, and made the decision to remove the possibility.
And for some reason they morphed into using the odd versions for the preferred releases (LTS), and even versions are dev releases.
They can spin it however they want, but I still think the jump from 1.x to 10, 11 etc. is a marketing ploy :)
Cliff,
Most of mine are running either on raspberry pi's or 1.86 GHz Atom boxes. The Atom is perfectly happy up to 20-25 conversations as long as you don't transcode, and most of my customers need far less than that.
Quote from: mdodds on April 15, 2015, 09:21:51 PM
And for some reason they morphed into using the odd versions for the preferred releases (LTS), and even versions are dev releases.
They can spin it however they want, but I still think the jump from 1.x to 10, 11 etc. is a marketing ploy :)
It is incorrect to characterize the even numbered versions as development versions. They are standard releases but not covered by long-term support. The odd-numbered development versions prior to 10 were never
released, they were only available as snapshots, obtained directly as checkout from the source control system.
Sorry unbeldi, you are correct. It's the server guy in me coming out 8) I consider anything that's not LTS to be a dev release, such as any Ubuntu release that's not an even year .04 version, even though I know they are really not dev versions. On a related note, have you used Asterisk 13?
Just got a Grandstream HT-502 in to use on another system and decided to test it for pulse dial. No workee :(
I wonder if I am missing a setting somewhere or if they have updated the firmware, or changed it internally and it doesn't work anymore.
Anyone using one successfully?
Quote from: mdodds on April 24, 2015, 02:20:05 PM
Just got a Grandstream HT-502 in to use on another system and decided to test it for pulse dial. No workee :(
I wonder if I am missing a setting somewhere or if they have updated the firmware, or changed it internally and it doesn't work anymore.
Anyone using one successfully?
It should work. What is the firmware version on yours?
Software Version: Program-- 1.0.5.10 Bootloader-- 1.0.0.15 Core-- 1.0.5.9 Base-- 1.0.5.10
This is not the latest version however, which I think is 1.0.13 or so.
Here are the versions:
HT-502 V2.0A
Software Version: Program -- 1.0.12.1 Bootloader -- 1.0.0.18 Core -- 1.0.12.1 Base -- 1.0.12.1
Extra -- 1.0.12.1 CPE -- 1.0.1.44
Mine is
Product Model: HT-502 V1.2A
Set the Preferred DTMF mode to SIP INFO, and disable DTMF negotiation.
Establish a call to the unit from another extension, or use a DTMF tone dialer with the rotary phone to call another extension.
While maintaining the call, watch the SIP messages when dialing a number on the phone during the call. You should see SIP INFO messages with the digit that was decoded from the dial action.
For example you should see a SIP message like this when dialing "6":
INFO sip:0666@10.201.239.202:5060 SIP/2.0
Via: SIP/2.0/UDP 10.201.239.171:5060;branch=z9hG4bK492069190;rport
From: "5021" <sip:5021@pia.example.org>;tag=1173897288
To: <sip:0666@pia.example.org>;tag=as6b896e56
Call-ID: 551528776-5060-10@BJJ.CAB.CDJ.BHB
CSeq: 104 INFO
Contact: <sip:5021@10.201.239.171:5060>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-502 V1.2A 1.0.5.10
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/dtmf-relay
Content-Length: 24
Signal=6
Duration=240
ALL of my calls are VoIP in one form or another, no matter where I am and many originate from Rotary dial phones. Luckily the 2 cities I dial most have no prefix overap so I have 7 digit dialing to fixed lines in 2 cities . There are some identical prefixes but that is where one is a cell phone and in Mexico cell phones are never dialed the same as fixed phones so it is a non-issue. (there are some cell phones in Mexixco with fixed dialing however)
I run X86 Vanilla Asterisk at home as a Cellular gateway using Chan_dongle and POTS gateway. This system is in my home and I do not register phones to it. It recives calls voia POTS and Cell (with Fixed dialing number) and sends to main server. Also handles many outbound calls
I have another Cellular Gateway running Vanilla Asterisk on X86 in city 2 no phones registered to it. it receives calls via incoming (fixed dialing) number and routes them to main server
I had a Raspberry Pi running Free PBX as a cellular gateway in a third city. No phones registered to it. This is now off line
I bring it all together on a Elastix server running as a VPS on a dedicated server which I rent and divide into PBXs on VMs for clients. One of those PBXs is my own
for rotary dial phones I have found no better ATA than the UTStarcom IAN-02EX . I have tried the grandstream HT503 (only very recently) but have issues with some rotary phones, particularly Ericssons. This is not an issue of dial speed rather a misread number with an extra pulse. I have unlocked Vonage Vportal only to find that it is very unreliable and often goes unregistered saying "cannot connect to Vonage network (rather my server)". I was told this may be from using both ports but it still seems the same with only one port. Although the Vportal does seem to accurately read the digits consistently, and displays them on display.
If you do look into a UTStarcom IAN-02EX (about 28 USD NEW here in Mexico) absolutely avoid unlocked units as you may only be able to use one port. I have used MANY rotary phones on these models and never had an issue , even when I had questionable dial speed.
So yes I have asterisk between my rotary phones and POTS lines as well as between my rotary phones and VoIP liines and between rotary phones and cell lines (that's a rare one).
I also use :
Google voice with Simonics.com as a go-between for USA number and outbound calling
FreePhoneline.ca for C anadian number and calling
messagenet.it for a UK number
VoIP.ms additional ported USA numbers and outbound calling
Rapidvox outbound provider
Flynumber.com for an Australian number, and Canadian number
All with Least Cost Routing because I am cheap (Cheap is good free is better, so Free is a very good price!)
NorthernTelcom if you need help with your configuration , call me
Mark
1 503 489 7870 USA
1 604 800 9818 Canada (soon to be migrated to a new number and not sure what that is at the moment)
Quote from: markosjal on December 08, 2016, 09:48:38 PM
If you do look into a UTStarcom IAN-02EX (about 28 USD NEW here in Mexico) absolutely avoid unlocked units as you may only be able to use one port.
Did you mean avoid *locked* units, not unlocked?
What I mean is any unit that was locked to a provider and is now "unlocked" . The UTStarcom from Lingo had bad firmware and only one port was usable.
Quote from: mdodds on April 24, 2015, 02:20:05 PM
Just got a Grandstream HT-502 in to use on another system and decided to test it for pulse dial. No workee :(
I wonder if I am missing a setting somewhere or if they have updated the firmware, or changed it internally and it doesn't work anymore.
Anyone using one successfully?
I use a Grandstream 502 and it works with pulse dialling. The first thing I did when I got it was to turn off any firmware updates in the settings. I've had it about 18 months now. The only problem I had was that it didn't work with certain UK phones. It turns out the make/break ratio of the UK dials can fall just outside the range the Grandstream will allow. A quick tweak of the dial contacts cures this though.
Hi all, I'm back after a fairly lengthy hiatus.
There seems to be 2 main hardware revisions for the 502, 1.2 and 2.0. I have not had any luck with 2.0 versions and pulse dialing. Unfortunately 1.2's are only available used. I did have success with my Digium IAXy ATA, but I only have one of those and they are discontinued now.
However, my Panasonic KX-T61610 just arrived yesterday so the pulse to DTMF problem is pretty much solved! :)
The Panassonic is definitelty the way to go. It's much more forgiving of slightly mis-set dial speeds etc than the Grandstream ATA.
For Grandstream users, I wonder if there is a way of rolling back the firmware?
Quote from: andy1702 on August 17, 2017, 03:49:05 AM
I use a Grandstream 502 and it works with pulse dialling. The first thing I did when I got it was to turn off any firmware updates in the settings. I've had it about 18 months now. The only problem I had was that it didn't work with certain UK phones. It turns out the make/break ratio of the UK dials can fall just outside the range the Grandstream will allow. A quick tweak of the dial contacts cures this though.
Similar to what I have seen with some Ericsson dials from Mexico as well. It is not the pulses per second it seems rather the duration of each pulse. Much depends on the model/age of the dial it is the newer ones that seem to be problematic, not the older ones.
http://www.dslreports.com/forum/r25525579-Looking-for-an-FXO-device-needs-pulse-dialing-support
I'm researching this as well, here is what I found. All Devices are Grandstream and this is not a complete list.
- HT502
HT70X (All in this series with updated firmware.
HT80X/81X with firmware 1.0.5.11 (New December 2017) or higher
The Following settings need changed on the ATA (dial plan assumes PBX):
- High Ring Power to Yes: This increases the available ring current
Enable Pulse Dial to Yes: This enables the Rotary Dial (you can leave this off for touch tone).
Dial Plan (Pulse/15xx Only): add the following in the front for every feature code with a * <11FC=*FC> | Where FC=digits of feature code. This will get complicated for speed dial and parking as speed dial would be <110SPD=*0SPD> |where SPD = Speed dial number (the entire dialed code must be specified). Also Remember the "|" as that is the OR keyword.
Expanding on kb3pxr's list
Grandstream
HT502
HT503 (on FXS port)
HT70X (All in this series with updated firmware.
HT80X/81X with firmware 1.0.5.11 (New December 2017) or higher
UTSTarcom
IAN-02EX
Motorola
VDV-21 (hacked Vonage unit, and possibly VDV-22)
The most reliable for properly detecting pulse dialing that I have seen to date is the UTStarcom IAN-02EX . Both Motorola and Grandstream units tested to date have exhibited issues on some phones. In have NEVER had an issue with the UTStarcom IAN-02EX
As of March 6, 2018, all current generation Grandstream devices with FXS ports will support Pulse dialing. The 500 series, current UCM models (DAHDI), and all devices with 8 or more ports support pulse dialing by default and do not require any configuration changes. All devices with 4 or more ports, except the HandyTone 704 support pulse dialing without additional configuration.
For Handytone devices in the 70x and 80x series Pulse dialing and High ring power are disabled (or not present) by default. To obtain this functionality you must upgrade the firmware to a compatible (ideally latest) version and select yes for "Enable High Ring Power" and "Enable Pulse Dial" on the respective interfaces in the configuration. It has also been reported that the HT802 does not disable pulse dialing capabilities during the call therefore having the effect of using a Pulse to Tone converter.
HT814 and HT818 in the Handytone series support pulse dialing by default and is not configurable, as of Firmware version 1.0.5.11 the HT814 also supports high ring power. Scant details are available on firmware 1.0.5.18 for the HT818.
All Grandstream devices with 16 or more ports seem to support both High ring power and pulse dialing without any configuration. Most Grandstream devices with 24 or more ports also support Neon message waiting lamps.
For gateway devices containing 16 or more ports, pulse dial support is usually hard to find documentation on and may be one sentence in the administration guide to the effect that Pulse dialing is supported. In the 16 port and higher range, additional features can be configured. Many devices start supporting additional Visual MWI modes (Neon and Polarity), feature code customization (enable, disable or change code), and even MultiFrequency (as in Blue Box) signaling. of course, these are enterprise grade devices and are priced accordingly.
I also mentioned DAHDI earlier. Many interface cards for Asterisk PBX systems use the DAHDI (Formerly zaptel) drivers and use similar hardware. Pulse dialing, high ring power, and various message waiting schemes can be configured. Neon is a bit tricky though. On DAHDI, you must select the HVAC (or neon alias) mode instead of HVDC. The output from the SLIC is in fact DC. If you read the source code (and the referenced technical document) you will find that the Neon (HVAC) mode turns on the ringing generator at 4 Hz for 1/8th of a second (half a cycle) to flash the neon light. These cards are not cheap, and the functionality of the cheap Chinese versions are unknown (My PBX is a Raspberry Pi 3).
Note about Grandstream's "Enable High Ring Power" setting: Grandstream devices that have, or later receive this setting operate their ringing generators at a reduced voltage presumably to save power. The hardware capability to run at the higher voltage is not removed and is later made available again in software. While this causes ringing difficulty with mechanical bells, this can be overcome on Western Electric C-type (I haven't tested P or M types) by changing the bias spring adjustment to the weak setting, I haven't had my Automatic Electric equipment connected in about a decade so I haven't tested it.